1. Field of the Invention
The present invention generally relates to wireless telecommunications, and more specifically, relates to a system and method for adaptively increasing the efficiency of audio data transmission over a communication network.
2. Description of the Related Art
Technology advancement has made mobile telephones or wireless communications devices inexpensive and accordingly ubiquitous. As wireless telecommunication devices are manufactured with greater processing ability and storage, they also become more versatile and incorporate many features including direct radio communication capability between two or a group of individual handsets. This direct radio communication capability is commonly known as the push-to-talk (PTT) or “walkie-talkie” feature that allows a user with one handset to communicate with the device of a predefined set of members of a group without dialing a destination telephone number.
In one version of a PTT system, a wireless telecommunication device, such as a handset, uses one single frequency for both upward and downward communications with a remote PTT server, while in a normal wireless communication a wireless telephone uses two frequencies for communicating with the PTT server, one for outgoing and one for incoming communications. The PTT system requires the person who is speaking to press a PTT button while talking and then release it when done. Any listener in the group can then press their button to respond. In this manner, the system determines which direction the signal travels. In a typical configuration, when a user makes a call to a receiving party or a group of receiving parties using the PTT system, the user's handset first makes a request to a remote server by informing the server it is ready to transmit. The remote PTT server verifies that no other party is using the communication channel and the channel is available then assigns the channel to the user. The user's message is received by the server and the server sends the message for each and every receiving party. After the message is transmitted to every receiving party, the channel is released and ready for use by other parties.
During the process described above, the audio from one user is sampled and digitized by a device, such as vocoder, at the handset. The digitized data is then assembled into frames and the frames are packed into data packets and transmitted over the air to the server. The server receives the data packets and sends them to their destinations. At their destination, the digitized data are extracted from the data packets, reassembled into audio streams, and played to the receiving party. In each data packet there may be one or more frames. Usually, the number of frames is set by the network standards or network service providers, and is independent of network conditions.
The network efficiency is increased when more frames are packed into a single data packet, and the efficiency is at its lowest when there is only one frame per data packet. However, incrementing the number of the frames per data packet also increases the delay at the user's handset. The handset has to wait for audio data from the user, assemble the audio data into multiple frames, and pack the multiple frames into one data packet before transmitting the data packet to the server.
Therefore, it is desirous to have an apparatus and method that enables a server to adaptively bundle different number of frames into a single data packet, such that the network efficiency is increased and it is to such apparatus and method the present invention is primarily directed.